Some more meditation: Lets consider call 200 -> 201. Phone 200 (192.0.2.254) gets 200Ok message (packet 21 in your trace). And according to sip standard it must send Ack message to address, specified in Record-Route header. But Record-Route header points to this phone itself. So... Ack message never goes out of the phone...
Problem is that sipxproxy sets wrong Record-route header (As I told you allready). How to heal it? It's more difficult question... But I think that you will want to find out the root cause of that ip address muddle... (Reinstall from scratch will not take much time... ;) ) Rgds, Nikolay. > -----Original Message----- > From: [email protected] > [mailto:[email protected]] On Behalf Of > Nikolay Kondratyev > Sent: Friday, February 04, 2011 3:02 PM > To: 'Discussion list for users of sipXecs software' > Subject: Re: [sipx-users] phone INVITE -> TRYING -> RINGING > -> connect?but no audio/voice with PolyCom SoundPoint 650 > > Ok, now I see all signalling messages in pcap files. > (unfortunately your xml trace file does not contain the whole > call...). > > I do not change my opinion: > You do not have voice because one of your polycom phones > (200) does not send Ack message. > Or to be precise: phone 201 sends 200 ok, but does not get > Ack, that’s why it (phone 201) does not send rtp stream, and > may even not play into it's handset received rtp stream. > > Regarding "phones talks directly to each other". > RTP stream is to be sent directly between the phones. > But all signaling (SIP) messages is to be sent through sipx. > Sip standard provides special header for that: Record-route. > > Lets consider your call 201 -> 200. You can hear voice, but I > still think that there is something GROSSLY wrong there. > Look at the packet 16 - Invite from sipx to 200. > It contains 3 via headers. > The first one (count from the bottom) is set by the originating phone. > The other two are set by sipxproxy process, and must contain > sipx address, i.e. 192.0.2.10., but there is 192.0.2.254, > which is polycom phone. > Moreover there is two record-route headers with 192.0.2.254 > address. This headers are set by sipxproxy process too. > Looks like sipxproxy process thinks, that it has address > 192.0.2.254. Which is GGRROOSSSSLLYY wrong. > > It may happen if you (by mistake) configured 192.0.2.254 > address somewhere in sipx configuration. > What is in /etc/sipxpbx/sipXproxy-config file? (Do not edit > it manually). > > Of course the above is just a guess, that I can make looking > at your trace... > > Rgds, > Nikolay. > > > > -----Original Message----- > > From: [email protected] > > [mailto:[email protected]] On Behalf Of Claas > > Hilbrecht > > Sent: Friday, February 04, 2011 1:52 PM > > To: Discussion list for users of sipXecs software > > Subject: Re: [sipx-users] phone INVITE -> TRYING -> RINGING > > -> connect? but no audio/voice with PolyCom SoundPoint 650 > > > > Hello Nikolay, > > > > thanks for the response. > > > > > I would guess that you do not have voice when calling > > 200->201 because > > > there is no Ack sent by "200-phone". And in the trace there > > is one way > > > RTP. I would say that you should have "one way audio", not > > "no audio". > > > I > > > > I omitted RTP in the trace at the first post to save space > but here is > > the full log. > > > > > don't use polycom phones and I can't advice if you use > appropriate > > > firmware version. > > > > The IRC (#sipxecs) folks told me to use this version. > > > > > I see another strange thing in the trace. Sipx always sets > > > Record-route header, pointing to itself, so that all messages go > > > through sipx. I do > > > > Hmm, maybe this is because of HD audio? Reading > > <http://wiki.sipfoundry.org/display/sipXecs/Polycom#Polycom-HD > Voice> it seems that the PolyCom phones talks directly to > each > other. > > > > > not see this header in the trace you provided, and I see > > Bye message > > > sent directly from one phone to another. That is bad. There is > > > something wrong in your setup. Search the wiki for topology > > > description and setup examples... > > > > I think my setup is the most simple one. I have LAN > > 192.0.2.0/24 dedicated to VoiP. DNS and DHCP are handeled > by sipXecs. > > A SmartNode 4638 is used to connect sipXecs to our PSTN lines. Only > > VoIP devices are connected to the lan. > > > > > Your traces does not show full message flow. > > > Did you take traces from the phones (mirrored ports)? > > > > For debugging cases like this one I use a plain old 100 > MBit/ HUB (yes > > a HUB, no switch). So all traffic is "mirroed" to all ports. Makes > > debugging with tcpdump/wireshark much more easier. But as I said > > before I use WireShark to filter only the call. > > > > > I'd better just run "tcpdump -s 0 -w filename.cap" at the > > sixp server > > > and then transfer filename.cap to your pc and view it with > > wireshark. > > > You'll see full message flow... Phone1 <-> sipx <-> phone2. > > (Of course > > > you'll see only packets that go through sipx). Or you may > > want to use > > > sipviewer > > > > > > http://wiki.sipfoundry.org/display/sipXecs/Display+SIP+message+flow+us > > > ing > > > +Sipviewer It will show interprocess sipx communication, > > which is very > > > useful for troubleshooting. > > > > Attached you will find a "no-audio.xml" for the sipXviewer. > > > > Thanks for your help > > > > Mit freundlichem Gruss > > Claas Hilbrecht > > > > -- > > http://www.linum.com mailto: [email protected] > Linum Software > > GmbH Langer Wall 5, 37574 Einbeck, Germany > > Tel: +49-5561-926730 Fax: +49-5561-926750 Handelsregister > > Amtsgericht Göttingen HRB 131128 Geschäftsführer > Claas-Jörg Hilbrecht > > > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/
